1. Field of the Invention
The present invention relates to a mobile telecommunication system and, more particularly, to formatting voice data in a mobile telecommunication system.
2. Background of the Related Art
FIG. 1 illustrates a configuration of a mobile telecommunication system in the related art. Referring to FIG. 1, when a mobile subscriber communicates voice data with a wire subscriber, the mobile telecommunication system compresses the voice call in a mobile terminal (MT) 1a–1n and transmits the compressed voice call to a base station controller (BSC) 3a–3n, via a base transceiver station (BTS) 2a–2n. The voice call can be compressed into digital data by a compression algorithm. The BSC 3a–3n converts the compressed digital data into pulse code modulation (PCM) data and transmits the PCM data to a mobile switching center (MSC 4. To convert the compressed digital data into PCM data, the BSC 3a–3n employs a vocoder. The MSC 4 can transmit the PCM data to a public switched telephone network (PSTN) 8, in which the corresponding wire subscriber is matched. The corresponding wire subscriber can be selected by a switching circuit.
So far, the method of providing a voice call generated by the MT 1a–1n of the mobile subscriber to the wire subscriber has been explained. Similarly, it is also possible to provide the voice call from the wire subscriber to the MT 1a–1n following the same procedure described above in an inverse order. In addition, to provide the voice call from MT 1a to another MT 1n, the same procedure can be applied, again.
In other words, the vocoder of the BSC 3a–3n can convert the compressed digital data from the MT 1a–1n to PCM data. Unfortunately however, as the vocodet repeatedly conducts the conversion process from the compressed digital data into the PCM data, it increases the quantization error as well, deteriorating the sound quality as a result.
As an attempt to solve the above problem, another method has been tried. For example, when the vocoder in the BSC 3a–3n should provides the voice call between mobile subscribers, the vocoder does not convert the compressed data from the BTS 2a–2n into PCM data to deliver it to the MSC 4. Instead, the vocoder sends the voice data to the MSC 4 after packetizing it to have a certain format. This operation can be carried out when a bypass mode is designated by a call process control of the BSC 3a–3n and the MSC 4. Also, if the vocoder at a receiver is in the bypass mode, the vocoder can recognize the data received from the MSC 4, through the communication time slot, as a voice packet data in the bypass format, not the PCM format. Then, the vocoder will decompose the data right away and send out a corresponding voice packet to the BTS.
FIG. 2 is a related art configuration of a vocoder inside of a BTS 2a–2n controller. The vocoder 20 includes a packet matcher 21, a voice coder/decoder 22, a bypass controller 23 and a PCM matcher 24. In the case of connecting the voice call between mobile subscribers with this vocoder, if the voice packet is transmitted to the packet matcher 21 from the BTS 2a–2n, the packet matcher 21 matches the voice packet and sends the matched voice packet to the voice coder/decoder 22, instead of the bypass controller 23.
The vocoder 22 is preferably pre-designated in the bypass mode. The bypass controller 23 transmits the voice packet to the time slot as it is. The voice packet transmitted to the time slot undergoes the packeting process in a special format and is sent out to the MSC 4 via the PCM matcher 24. Data received from the MSC 4 proceeds in a reverse order, of the aforementioned transmitting procedure for the voice packet, and is sent to the bypass controller 23. Later, the vocoder recognizes the data as the voice packet data, decomposes the data, and finally sends the voice packet to the BTS.
FIG. 3 illustrates a data configuration of a bypass packet of the vocoder in the related art. The bypass packet format comprises up to 40 bytes of data in total, including a maximum 32-byte encoding packet data 33 received from the BTS 2a–2n, a 4-byte message 32 for transmitting a signal, and a 4-byte preamble 31 for distinguishing a final/ending message. Additionally, the bypass packet may include a 290-bit dummy 34 and 30-bit cyclic-redundancy-code (CRC) 35. Thus, the packet data format can include up to a total of 80-bytes of data, and the packet data format can go through the same procedure one more time before it is transmitted. After being formatted, the bypass packet is sent to the MSC 4.
On the other hand, when a synchronous system is involved, 32-byte encoding packet data 33 is transmitted and received every 20 ms between the BTS 2a–2n and the BSC 3a–3n. Further, between the BSC 3a–3n and the MSC 4, a 1-byte bypass packet every 125 us (i.e., a 160-byte bypass packet every 20 ms) can be transmitted and received.
FIG. 4 illustrates an operational procedure of the bypass mode. The formation of the bypass packet format includes adding the 4-byte message 32 on the basis of the 32-byte encoding packet data 33 (S41) and further adding the 4-byte preamble 31 (S42). In addition, after adding the dummy 34 and the CRC 35 to the encoding packet data 33, the message 32 and the preamble 31 (S43 and S44), the adding process (S41–S44) is repeated entirely to generate a second block of data with the same length (i.e., 80 bytes) (S45). Together, the two blocks of data complete the final bypass packet format (160 bytes). A final bypass packet of this form is delivered to the MSC 4, by communicating 1 byte every 125 us (S46). Meanwhile, the packet data received from the MSC 4 is decomposed in a reverse order of the above-described procedure, and the encoding packet data 33 is extracted from the packet data format and sent to the BTS 2a–2n. 
The most typical and generic data length that is effective in the related art is within a range of 32 to 36 bytes. If the preamble is added, the maximum continued data length reaches 40 bytes. Accordingly, the voice data used in practice is approximately one fourth of the total 160 bytes. Therefore, data resources are often wasted. Moreover, during the repetition of the packet data format operation, it is always possible that 80-byte data of the first half frame 30a, in FIG. 3, can include an error. In this case, the BTS 2a–2n has to resend the correct data. That is to say, if the first half frame 30a or the second half frame 30b of the 80-byte packet data format contains an error, in a particular bit, it is regarded as a CRC error and the BSC 3a–3n recognizes the data as being invalid. Thereby, information on the error occurrence or a useful and effective handling method are precluded.
In addition, in terms of the properties of the voice data in the real time mode, no band width assigned for resending the voice data in the next 20 ms frame. Although it seems possible to regenerate and send the regenerated packet data frame, in practice the entire frame (30a and 30b) can be at risk of losing validity for even 1-bit error. Even worse, there is no way to correct the error.
If the preamble 31, identifying a start point of the packet data, has an error during the bypass mode of operation, the valid start point is nowhere to be found. Also, when combining the preamble 31 with the encoding packet data 33, the encoding packet might have its own preamble as well. In such a case, it is impossible to find the start point of the frame.
If, by any chance, a 160-byte frame is lost, not only is the voice data no longer valid but also a synchronization with the next frame is no longer valid. The synchronization loss requires a considerable amount of time to overcome, when it occurs for a large number of frames.
Therefore, the related art is very disadvantageous in that it fails to accomplish an original goal of improving the quality of sound, during the voice call between mobile terminals, by reducing the quantization procedure.
The above references are incorporated by reference herein where appropriate for appropriate teachings of additional or alternative details, features and/or technical background.